Variable delay line signal processor for sound reproduction

ABSTRACT

An electrical delay line which has variable delay controlled by a signal input thereto is connected in a sound signal channel for signals such as human speech, to compress or expand the sound signal waveform depending on whether the time delay in the line is increased or decreased. By periodically sweeping the delay line from minimum to maximum time delay or vice-versa, repeated segments of a continuous sound signal waveform are processed so that an output audio signal can be obtained having the original frequency components of the signal and occupying a time duration which is equal to or smaller or larger than the original sound sequence with the successive segments of the signal processed by the variable delay line assembled with regard both to the significant parameters of human speech or other coding and the electrical conditions imposed by the system to produce a composite audio signal which is an intelligible replica of the original and substantially free of annoying aberrations introduced by the delay line processor. Variable delay using analog or digital signal storage is also provided.

United States Patent [191 Schiifman 1 1 VARIABLE DELAY LINE SIGNALPROCESSOR FOR SOUND REPRODUCTION [75] Inventor: Murray M. Schiffman,Newton,

Mass.

[73] Assignees: Cambridge Research and Development Group, Westport,Conn.; Sanford D. Gr'eenberg, Washington, DC; D. T. LiquidatingPartnership, New York, NY. 22 Filed: Aug. 13, 1971 [21] Appl. No.:171,571

[52] US. Cl. 179/1555 T [51] Int. Cl. G101 1/06 [58] Field of Search179/15.55 T, 15.55 R, 179/15 BW; 178/54 HE, 66 TC, DIG. 3

[56] References Cited UNITED STATES PATENTS 3,202,769 8/1965 Coleman,l78/5.4 HE 3,093,796 6/1963 Westerfield.. 179/1555 R 3,278,907 10/1966Barry 179/1555 T 3,409,736 111/1968 Hurst 178/6.6 TC

Primary Examiner-Kathleen H. Claffy Assistant Examiner-Jon BradfordLeaheey Anor'ney'Charls E. Pfund, Esq; and Chittick, Thompson & Pfund 1Jan. 15, 1974 [57] ABSTRACT An electrical delay line whichhas variabledelay controlled by a signal input thereto is connected in a soundsignal channel for signals such as human speech, to compress or expandthe sound signal waveform depending on whether the time delay in theline is increased or decreased. By periodically sweeping the delay linefrom minimum to maximum time delay or vice-versa, repeated segments of acontinuous sound signal waveform are processed so that an output audiosignal can be obtained having the original frequency components of thesignal and occupying a time duration which is equal to or smaller orlarger than the original sound sequence: with the successive segments ofthe signal processed by the variable delay line assembled with regardboth to the significant parameters of human speech or other coding andthe electrical conditions imposed by the system to produce a compositeaudio signal which is an intelligible replica of the original andsubstantially free of annoying aberrations introduced by the delay lineprocessor. Variable delay using analog or digital signal storage is alsoprovided.

65 Claims, 64 Drawing Figures Audio 53 N SpeechBand A f e VariableBlanking 335 5OOO Output 5| P ypok Input l l Dem 1 Circuit a BQndP jevlce A Te I Band-Pass 7| j k Filter 74 Filter 76 9 l 72 F" 73 I 57 GaplElv 75.

c\ |B| E lBlonking Gate Rompevel Blanking 6? Pulse Amplitude ChangerGenerator I I i 61 59 U. u.

Comp Sample 63 J\. Period Ramp Pulse- Train Generator 64 66 PATENTEB JAN15 1874 t for =4 Delay for 6:3, d :g

l t for e-4 Delay for c- 4 d SHEET 02 0F 10 T Input Chunk Oufput SampleOUT 7 m 11 W /Z ?or (2 1 [gno (Cor recred Signal) EL 7 H C-T 1 FIG. 2(a)SIGNAL COMPRESSION i mm.

' T Somple Period=cT T I 1 C m ou'r I e=%, T Input Chuhk=eT T foufputChunk,

Sample Period t'=tT c'=c--1= ILLUSTRATION FOR 9 =4 1e 11 et=t for e =1 1n Qa-T (Corrected Signal) 1 FIG. 2(b) SIGNAL 4'3 EXPANSION e z; me I g &""m maummm rzclhjnj TOUT) INVENTOR 1 i MURRAY M. SCHIFFMAN our TO'UT BYet =t' Sample Period =eT =T ATTORNEY PATENTEUJAN 15 m4 SHEET DJUF 10INVENTOR MURRAY M SCHIFFMAN QMeTGM ATTORNEY PATENTEDJAN 15 m4 SHEET 0%OF 10 5150 85m A: Emmi .rDnTrDO mmkfm ozikoozw AB .rDnCbO omEmwQ ASINVENTOR MURRAY M. SCHIFFMAN WSGM- ATTORNEY Pmmmmsmu 3.786.195

sum user 10 Avc At Speech Bond Audio 51 Playback Inpui Amplifier\lgriloble 5:1 5:9 333 -sooo 1)) eu r De e Vorroble Line 7l AmplifierBorEHPoss 74 Bond-Puss 5 er FiHer 7 r 2 r 57\ Gop lHer 75 C\ B E f RompLevel BOnking I 6K eei e'izr r Changer Inverter P62 Exp. Comp. Sample 763 J\\ Period Romp Samp Pulse- Period T ain 1 65 Cnil Generator \64 56Romp Pulse Generator 84 B Speech 8 Sig no! r i b) W I I Chunk LenqihSample Period gup (c) B E |npuf EXPANSION CHUNK Output (d) J INVENTOR Av G p MURRAY M. SCHIFFMAN maew ATTORNEY mm 15 m4 3388.195

SHEET UYUF 10 53 Audio In Out FIG. l2

*1 B "-+1 F|G.|3 (b) BLANKING "COMPRESSION FIG. I4

EXPANSION K INVENTOR F v MURRAYMSCHIFFMAN ATTORNEY mimmm 15 m4 3786L195sum use; 10

I FIG. l5 1 w 5 51 Fl F1 (e) B hf) I H H E n m INPUT SIGNAL 2 AsRi ASR2ASRN OUTPUTHZIGNAL FIG. I? r 1 SHIFT n3 FREQUENCY ILJ'LF GENERATOR n4IELFL T INVENTOR MURRAY M. SCHIFFMAN we". GM

ATTOR N EY VARIABLE DELAY LINE SIGNAL PROCESSOR FOR SOUND REPRODUCTIONBACKGROUND OF THE INVENTION The field of this invention is theprocessing of human speech signals or similar signals for ultimatecomprehension by the human listener at substantially the natural ornormal frequency component distribution but at time interval durationswhich are usually different from the original time duration of thespeech utterance.

Sound compression and expansion systems which utilize relative motionbetween a magnetic tape record medium and the air gap of the pick-uphead which senses the recorded signal on the magnetic medium are wellknown as exemplified by the patent to Schiiller 2,352,023. Devices ofthis type suffer from the usual operational, cost and weight limitationsinvolved in equipment which utilizes substantial mechanical motioncomponents. A delay line version of time compression and expansion forreal time signals is also well known as shown, for example, in thepatent to French et al. 1,671,151, where a voice signal is propagatedalong a delay line and a movable pick-up repeatedly scans the delay lineto sense the signal propagating therethrough with the relative velocitybetween the pick-up head and the propagation velocity of the wave in themedium giving bandwidth compression or expansion for the purpose oftransmitting the signal over a narrow band telephone line. Later workersin the field eliminated the mechanical motion portions of systems suchas French et al. by substituting electronic switching sequentially alongtaps on the electrical delay line thereby providing frequencycompression or expansion of thesignal for transmission over a narrowband line. The application of the sequentially scanned tapped delay linefor modifying the time duration of recorded speech signals withoutaltering the frequency components thereof is disclosed by Greenberg etal. U.S. Pat. No. 3,480,737. By relating the speed of scan of the delayline to the velocity of propagation in the delay medium and the relativespeed of the reproduction of a recorded message compared to the speechutterance from which it originated Greenberg et al. achieve timeexpansion or compression of a recorded speech signal without alteringthe frequency components thereof.

Another form of frequency-time transformation is known in the art usinga signal controlled variable time delay line for error correction.Systems of this type detect an unwanted frequency effect due to timeirregularities in the pulse'train of a repetitive signal or suchvariations in an audio system where the speed of the record medium pastthe pick-up head is subject to periodic variations which result in theproduction of the audible irregularity known as wow. In reproducing theoriginal signal these systems eliminate speed errors by servo-control ofthe time delay of a delay line which is interposed in the signalchannel. Audio systems which employ a reference or timing signal trackwith variable delay line compensation for playback speed are shown forexample in FIG. 9 of Coleman, Jr. U.S. Pat. No. 3,202,769 who shows anopen loop servo. Woodruff U.S. Pat. No. 3,347,997 shows a closed loopservo which adjusts playback speed to compensate for the relatively lowfrequency wow component whereas imperfections of a high frequency nature(i.e., flutter") are compensated by a variable delay line. Suchcompensation systems depend for their operation on the repetitivecharacter of the error signal and thus by The systems described in the{patents to Schuller,

French et al. and Greenberg et al., when used to reduce the frequency ofa speech signal, while compressing the time in which a given segment ofspeech is reproduced, inevitably involve discarding a portion of theoriginal speech wave. The ratio of the speech signal discarded to thatwhich is used is directly related to the compression ratio and thediscard loss is inherently and fundamentally related to this process ofreducing the frequency and compressing the time for the processing of agiven passage of speech. Since the portion of the speech which isreproduced alternates with portions which are discarded the problem ofmerging to reproduce sections in continuous time slots presents someproblem and various solutions have been offered.

Thus Schuller suggests a skewed air gap in his rotating magnetic pick-uphead or skewed tape approach to the point of contact with the rotatingair gap so that the arrival and departure of the magnetic tape recordrelative to the air gap will occur gradually as the skew provides atransition from zero to full air gap contact with the recording medium.

The patent to French at al. suggests a number of alternatives includingthe use of' two spaced transducers rotating in unison with respect tothe delay medium such that the message reproduced in compressed time inone transducer has superposed thereon from the other transducer themessage which would ordinarily BRIEF SUMMARY OF THE INVENTION The 1present invention provides compressionexpansion systems for speech orother coded signals in which the active frequency-time conversionelement is a signal responsive delay line which is directly interposedin the path between the signal source and the ultimate reproducer orutilization device which receives the converted speech wave. Such asystem does not avoid the inherent problem in time compressionoccasioned by the discontinuity resulting from discarding alternateportions of the speech wave. The discard portion of the speech wave maybe stored or cancelled in the delay line ordiverted from entering theinput terminal of the delay line which is directly in the signalchannel. In any case this signal and transients produced by lineswitching must be discarded as the variable delay line is repetitivelycontrolled by the delay signal between its minimum and maximum delayvalues. Line switching and discard occur coincidentally with therequirement for making contiguous two originally spaced portions of thespeech wave and hence both of these functions are accomplished by anumber of alternative embodiments herein disclosed which function in amanner consistent with requirements imposed by the parameters of thespeech signal itself.

Accordingly, it is the principal object of the present invention toprovide a speech compression-expansion system which utilizes a signalcontrolled variable delay line located directly in the signal channelbetween the signal source and sound reproducer which delay line isrepeatedly sequenced betweenmaximum and minimum delay values to modifythe frequency-time characteristic of the sound reproduced from theoriginal signal.

It is a feature of the invention to control the input and outputbandwidth of the system for the speech frequencies which are required tobe reproduced for intelligibility in relation to the compression ratiothereby to exclude those frequencies which would produce distortion andintermodulation due to insufficient sampling rate or excessive phaseshift per delay stage for the high frequencies and inadequate outputchunk length to attain frequency conversion for the lowest frequencies.

A further feature of the invention provides maximum discard intervals asdetermined by the maximum delay line length which in relation to thecompression ratio limits the actual original speech message discard to avalue which minimizes the loss of significant cues or transitions in thespeech code thereby minimizing the loss of information contenttransferred to the listener.

, A still further feature of the present invention is to provide, in asystem utilizing variable delay line speech compression or expansion,for signal processing at the point of juncture of two reproduced speechportions to suppress distracting noise components and also to avoid theintroduction of false cues which could modify the information conveyedin the subsequent speech segment. To this end, if required to suppresssuch noise, the transition between successive reproduced speech samplescan be modified by simple transfer function selection or control, orthetransition can be eased by the introduction of synthetic orspeechderived signal portions to approximate a smooth transition withina time interval which does not lose actual cues and under suchconditions that do not introduce false cues.

These and other features and advantages of the invention will beapparent from the following detailed description taken in conjunctionwith the accompanying drawings.

BRIEF DESCRIPTION OF THE DRAWINGS FIGS. 4(a) to 4(f) show waveformsuseful in describing forms of processing of a transition betweenadjacent reproduced speech samples.

FIGS. 5(a) to 5(d) show a set of curves representing active processingof the transition between adjacent samples.

FIGS. 6(a) to 6(e) show waveforms useful in describing the use of twodelay lines to effect transition between adjacent speech samples.

FIG. 7 shows a block diagram of a speech compressor-expander system inaccordance with the invention.

FIGSTSKa) 6823) show waveforms useful in describing the operation of thesystem of FIG. 7.

FIG. 9 shows a block diagram of a dual delay line system in accordancewith the invention.

FIGS. 10(a) to 10(d) show waveforms useful in describing the operationof the system of FIG. 9 for compression.

FIGS. l1(a) to 11(d) show waveforms useful in describing the operationof the system of FIG. 9 for expansion.

FIG. 12 is a partial block diagram of a modification. I IYG STIKGTEEISYE) show waveforms useful in describing the operation of circuit ofFIG. 12 for compression.

FIGS. 14(a) to l4(c) show waveforms useful in describing the operationof the modification of FIG. 12 for expansion.

FIG. 15 is a partial diagram of a dual delay line binaural system.

FIGS. 16(a) to 16(f) show waveforms useful in describing the operationof the system of FIG. 15.

FIG. 17 is a partial block diagram of a speech processor in accordancewith the invention using an analog shift register at the variable delayelement.

FIG. 18 is a block diagram showing gap filling with signal continuity ina system similar to FIG. 17.

FIG. 19 is a partial showing of an embodiment of the invention withvariable delay provided by an r-bit parallel digital shift register.

FIG. 20 shows an embodiment of the invention with variable delayprovided by a serial digital shift register.

FIG. 21 shows an embodiment using an analog storage memory matrix forvariable delay.

' FIG. 22 shows an embodiment using an r-bit digital random accessmemory.

FIG. 23 shows a logic diagram of a directional zero signal level gatingcontrol in a dual line system. 7

FIG. 24 shows waveforms useful in describing the operation of thecircuit of FIG. 23.

FIG. 25 shows graphically the clock frequency and maximum signalfrequency for the system of FIG. 17.

DESCRIPTION OF THE PREFERRED EMBODIMENTS The description of thepreferred embodiments will be preceded by a discussion of the parametersof the speech signal particularly as they relate to speech compressionfor reproducing a given speech message in a shorter period of time.Because of the fundamental and unavoidable limitations involved inspeech compression, the following discussion will proceed with primaryattention directed to the method and apparatus used in compression mode.Compression mode operation in the time domain results in the discard ofa fraction of the original information directly proportional to thecompression factor which is also the factor by which the time to presenta given speech sequence is decreased. The method and apparatus areusable however in expansion mode and the considerations involved for useof reproduced signals which occupy a greater length of time than theoriginal speech utterance will be described separately hereinafterqThesystem is also capable of frequency transformation without acorresponding time change to achieve a desired frequency signal, such asmay be involved in generating speech in amedium having a propagationvelocity different than air.

Referring tdF IGTi, a partiedar yaaauragqaaay line which providesmaximum time delay of 6ms for the final portion of the sample is shown.Assuming that the speech signal to be processed is limited to frequencycomponents between 333 and 5,000 Hz, certain parameters of the playbacksystem for compression can be defined. A magnetic tape 21 has the speechsignal recorded thereon of which the lowest frequency component at 333l-Iz'is depicted by sine wave 22 with the tape being drawn past apick-up transducer as it is wound on a take-up reel 24 at speed S. Theelectrical signal produced by the transducer 23 passes through acompression processor 25 and is reproduced as an audible signal fromspeaker 26.

The system 23 26 shown on line (a) of FIG. 1 just described reproducesthe recorded signal on the tape 21 without frequency or time change ifthe speed of takeup reel 24 draws the tape past the transducer 23 at thespeed of recording S and for this condition processor 25 would introducea fixed constant delay time of any value. Thus in line (b) of FIG. 1where c=1, the reproduction of the 333 Hz sinusoidal signal withoutchange other than fixed phase delay (which has been ignored) is shown.

speed is discarded and on line (c) is labelled discard."

This discard includes cycles 5, 6, 7 and 8 of the original wave 22 andrepresents the gap in the information content between successive chunkswhich are reproduced as audible signals. This audible output isrepresented on line (d) where the chunk is depicted as a piece of tape31 played at speed 2S and containing cycles 14 which afterprocessing iseffectively stretched into a piece of tape 32 occupying the originall2ms of recorded time and containing the cycles 1-4 at their originalrecorded frequency. In line ((1) it will be noted that the next cyclereproduced is cycle number 9 of the original wave after cycles 5-8inclusive have been discarded. The representation in line (d) of asmooth transition between the end of cycle 4 and the start of cycle 9should not be taken as representative of real signal conditions as wouldbe obvious from a consideration of an actual signal as opposed to theidealized signals presented in FIG. 1.

Lines (f), (g) and (h) of FIG. 1 illustrate the situation which prevailswhen the compression ratio is equal to five with the tape speed drawnpast transducer 23 at five times the recorded speed- S. With a finalsignal delay of 6ms maximum this compression ratio results in a chunklength of 1.5 containing 2 a cycles of the 333 Hz wave 25 of line (a)and again a discard interval of 6 ms equal to the final signal delay andcorresponding to a delay line length of 10rns at the end of the sample.The information gap, however, has increased to the point where the lasthalf of cycle number 3 and first half of cycle number 13 and all theintervening information in the original recorded wave have been lost inthe discard and this gap in the message represents 30ms of theoriginally recorded speech utterance.

The relations among the parameters of a speech compression system andthose pertaining to the information content of the speech codinginterr'elate in a manner to specify optimum conditions and place outsidelimits on the mode of operation of the systems of the invention for agiven intelligibility factor. These parameters can be examined withrespect to a particular system for various compression ratios and forthis purpose the system parameters for a system using a delay line witha maximum final signal delay AT of 6ms are set forth in the followingtable. i

i TABLE 1 Typical Parameters for Speech Compressor Chunk/Discard RatioComp Line (Playback (Recording Sample Rep Cycles Ratio Length cTime)Time Period Rate Sample m u/ M, nIt nnz T U c d (ms) (ms) (ms) llP. Hz)

1.25 2/9 6 24/6 30/7.5 30 33.3 10 1.5 2/5 7 1/5 12/6 18/9 18 55.6 v 6 235 8 6/6 12/12 12 83.3 4 3 l 9 3/6 9/18 9 111 3 4 6/5 9 3/5 2/6 8/24 s125 2 "a 5 4/3 10 1.5/6 7.5/30 7 A 113 2 pying 24ms of recorded time.This retained portion is designated a chunk and is shown in speeded formprior to processing on line (c) of FIG. 1 to include cycles number 1, 2,3 and 4. By virtue of the compression process and since the maximumfinalsignal delay is- The basis for the frequency-time transformationemployed in the present invention can be derived as follows. Consider asine wave V=E sin wt recorded with a tape recorder. If the tape isplayed back at 0 times the original recording rate, the result isV=Esincmt i where c is called the compression ratio. If c l the time iscompressed for any given speech passage and if c l time is expanded bythe factor e where e 1/0.

If the signal is then applied to a delay line in which the delay of theline is caused to increase linearly with time at the rate d, so as tocause the average delay of the signal to be c which represents the delayany point on the waveform will experience in passing through the line,then the signal (1) becomes V=E sin (c-c') wt The original signal isrestored if the delay is ct flo restore) (C such that c d/2 (c l), theaverage delay rate of the line 4 which is derived as one half the sum ofthe final and initial delay values of the delay line, which whenmultiplied by time, t, thus yields the total delay, ct;

FIG. 2(a) shows a plot for a given signal sample of signal output time,r vs. the corresponding input time, t Thus a line with a slope of 4represents signal of four times the original frequency or speed ofpresentation, and one-fourth the periodicity while a line with a slopeof 1 represents the resultant restored, or unchanged signal. In order toconvert such a signal (as represented by the line I of slope c 4) to onecorresponding to line II with a slope of 1, and with a correspondingfrequency decrease, it is necessary to increasingly delay the inputsignal, ct by an amount ct [or (cl)t] as shown at line III. Thus asignal chunk, T 24 has an ordinate which intersects III at the ordinatevalue c'T and this value when added to the time abscissa value at thepoint cT on I, delays the signal to T on line II. The delay dtintroduced by the delay line is shown by line IV. Such a delay line hasthe effect of delaying the instantaneous signal, t through a linearlyincreasing amount d-t for the interval from t to t as shown by line IV.Thus as in the case of the end signal at time t=T one-half the sum ofthe initial delay, ([1],, and the final delay, dT, yields an averagedelay value on line IV of c'T the amount required for restoration.

In more general terms the restoration may be achieved by cumulativelydelaying an input singal, t,,,, by an amount from which we obtain (4) InFIG. 2(b) the corresponding relationsfor signal expansion are shown.Line l with a slope of one-fourth represents a signal of one-fourth theoriginal frequency or speed of presentation. In order to convert such asignal to one corresponding to line II with a slope of 1, and with acorresponding frequency increase, it is necessary to decreasingly delaythe input signal, ct (=(1/e) t (1/4) t,-,,) by an amount ct l-e/e) t=-34!) from an initial delay of c'T This amount of delay, ct, at anypoint shifts the signal to the corresponding ordinate value on line II.The delay dt introduced by the delay line is shown by line IV Such adelay line has the effect of delaying the instantaneous signal by alinearly decreasing amount d't' for the interval from t to t as shown byline W Thus as in the case of the initial signal at time, t=0, one halfthe sum of the initial delay, d-T,,, and its final delay, dT yields anaverage delay value on line IV, of -c'T The process of linearlyincreasing time delay cannot continue indefinitely, and from time totime the delay line must be returned to its original length. If thisprocess is repeated at periodic intervals, provided the interval islonger than the period of the lowest frequency component of the signal,chunks of the original signal will be played back at the angularfrequency (c c')w and the rest discarded. When (3) and (4) aresatisfied, the system operates as though sections were cut out of theoriginal tape, pasted together, and played back at normal speed. Thesections of signal are heard at the correctfrequency but the informationis transmitted in a shorter time (if c l). The speech has beencompressed to 1/0 of its original length.

The values set forth in Table I have been plotted in FIG. 3(a). For anygiven compression ratio the sample time is given by the curve T and thechunk length is shown by the curve T The difference between these twocurves is the discard which is equal to the final delay to the signal atthe end of the sample period (6ms in the example shown in FIG. (3a).Entering the curve at any compression ratio, such as c 5 in FIG. (3a),one obtains the chunk and discard times for the tape running at c timesthe recorded speed and these values projected to the time axis show theactual original recorded time for the respective chunk and discardportions. As indicated for c 5 the chunk is 1.5 ms long and the discardis 6 ms long representing respectively 7.5 ms of recorded and reproducedinformation and 30 ms of discarded information. This latter value isrepresented by the quantity c AT which is also plotted in FIG. 3(a) Fora speech signal in which the lowest frequency 333I-Iz has a period of 3ms, a chunk length of 1.5 ms at c 5 corresponding to 7.5 ms of recordedtime will contain 2.5 cycles of the 333I-I signal. For any higherfrequency components in the speech signal more cycles will be containedin the 1.5 ms chunk. The length of the chunk should exceed the period ofthe lowest frequency to be passed (i.e., should include at least a fullcycle) otherwise satisfactory compression will not be obtained. Asindicated in FIG. 3(a) below the time axis at 3ms, the 333Hz signal isprocessed at sample periods approaching 3 ms would with its samplesreassembled accordingly produce a compressed output of poor qualitysince the sampling would then be causing a disruptive discontinuity fornearly every cycle of the 333 Hz signal processed. Sample periods lessthan 3ms would not permit completion of any one cycle so that theresultant reassembled output would not only contain the said disruptionbut would also begin to exhibit a basic change in its frequencycharacteristic in the form of waveform compression by truncation toproduce false frequencies. While this condition does not represent areal condition for a speech wave due to the complexity of the waveforms,this principle is controlling and sample periods less than the period ofthe lowest frequency wave in the speech signal will not provide propercompression.

Sample periods greater than the period of the lowest frequency wave willproduce compression and an interval of disruption exists from the regionwhere the sample period is only slightly greater than the period of thelowest frequency wave as indicated on the time axis between 3 ms and 6ms in FIG. 3(a). The result obtained within this period of disruption isa distorted expanded wave in which the effect of disjunctions betweensamples becomes extremely severe as the single cycle point is approachedand diminishes as the number of cycles in the sample increases. As apractical matter two and one-half cycles per sample is indicated as thedesired limit in FIG. 3(a) but in general the more cycles in the samplethe less the disturbance factor.

In order. to avoid the extreme distortion produced by waves which have alonger wavelength than the sample period, these lower frequencies shouldbe filtered out before the speech signals enters the delay lineotherwise these disjointed and highly distorted waves will be propagateddown the line and intermodulate with the desired signal and may severlydegrade the system performance. v

For lower values of the compression ratio than c 5, and keeping AT 6ms,the chunk length increases with the result that the actual time sampleincreases to greater than 7.5 ms and therefore more than the minimumnumber of cycles for the lowest frequency wave component will be presentin the chunk. Thus it would be at the users option to operate the lineover less than the 6ms indicated delay for AT to reduce the amount ofdiscard.

Considering the discard portion of the sample as a constant 6ms long atthe compressed rate of playback, the actual information loss is thecompression ratio times 6ms so that with c 5 the actual informationdiscarded for each sample is 30ms of recorded time. As shown on the timeaxis of FIG. 2 this is the interval from 7.5 ms to 37.5 ms and therelation of this loss of information to the intelligibility of thereproduced speech signal must be examined.

In general, human speech represents an extremely complex coding of arelatively limited set of sounds called phonemes which taken in contextwith the various attributes of the speech code such as thevoicedunvoiced components, pitch, formant frequencies and the continuumof sound pattern represented by sound energy (and the absence thereof)connected by the all important transititions between the temporalcomponents thereof constitutes an acoustic stream of infinite varietyand versatility. The ability of the human ear to receive this acousticmessage and the ear-brain system to decode the message is not altogetherunderstood since it appears that the readily comprehended informationrate far exceeds the mere acoustic response characteristics of the earas a receiver.

Fortunately, the ability of the ear-brain system to comprehend themessage which is conveyed by human speech signals is sufficiently goodto permit large portions of the actual acoustic stream to be lost ordiscarded without significant loss in the perception and comprehensionof the message information content of the acoustic signal. Since thecomprehension of message content decreases more rapidly than the recognition of individual words as the message is presented to the listener atincreasing rate,-the problem associated with the discard of a portion ofthe signal stream can be resolved in favor of comprehension and short ofthe point where intelligibility of individual words deteriorates. Thislatter point is reached where the loss or alteration of transitions orother ones representing the connection between a consonant and vowelsound results effectively from the discard of much or all of a given cueor cues so as to alter the apparent information content of cotiguousconcatenated chunks. Even before the point of absolute loss ofintelligibility is reached the limit of tolerance due to discomfort forsustained listening occurs as a result of the unnatural sounds and thefatigue which develops in the intense concentration required inattempting to extract the information content in the presence ofexcessive time clipping.

For the purpose of speech compression the loss of intelligibility can beassociated with discarding portions of the message containingsignificant cues or phonemes which components vary in length with theshortest being approximately l0ms to 20ms long. These short cues do notdominate speech but occur with sufficient regularity to make theirsystematic loss undesirable and hence a desirable upper limit for thediscard period would be considered to be 30 ms and preferably closer tol5ms. With this limit set for intelligibility of the reproducedsyllables and words the rate of presenting a given message can beincreased to the comprehension limit for any given listener and degreeof difficulty of the subject matter with minimum concern for-thelimitation which would be imposed by permitting loss or distortion ofthe word content or the generation of false cues from the concatenatedmessage chunks.- FIG. 3(a) indicates the recording time discard relationto compression ratio as the linear function cAT with the range from 15ms to 30 ms designated the discard uncertainty range. Thus the 6msdiscard at c 5 projects to include the real time recording interval fromtime t 7.5 to t= 37.5 which approaches the upper limit permitted fordiscard without undue loss of intelligibility as required not tocontribute significantly to the loss of comprehension in the messageperceived. Smaller values of c result in smaller actual discard time andhence the intelligibility is improved especially for those cues whichare at the lower end of the time scale, i.e., in the neighborhood oflms.

While Table I and FIG. 3(a) represent parameters for a typical speechcompression system having a final signal delay of 6ms and define thelimits of operation within fairly narrow limits, it will be appreciatedthat the principles involved can be adapted for use over a wider rangeof operation. Thus the variation of the actual frequency band of thespeech signal and the maximum length of the delay line are bothimportant design factors which influence the selection of thechunk-todiscard ratio and sample period for a given range of thecompression ratio c. On the other hand, the actual frequency range ofthe signal has an important bearing on the design of the delay linewhich must accommodate the frequency spectrum present in the signal asto such quantitative and qualitative factors as the voice pitch, the.presence of all or only some of the format frequencies for anindividual voice and the width of the signal spectrum over which linearphase-frequency properties must be preserved. The ultimate system usedhowever will embody design choices of the factors involved within thebroad limits herein defined.

FIG. 3(b) is a plot of corresponding relations for signal expansionshowing the initial gap, output chunk and maximum delay line lengthvariation with expansion ratio e for a given input sample interval T Theoutput gap occurs at the start of each sample period and thereafter forthe balance of the sample period the reduced frequency time-expandedoutput chunk appears. The maximum delay d T required is also shown as afunction of the expansion ratio e.

One aspect of the speech compression system described in connection withFIG. 1 has not been treated, namely, the audible output of thetransducer 26 when the variable delay processing unit 25 is switchedfrom maximum to minimum delay at the end of the sample period. Justprior to switching the delay line is loaded with the speech signal whichis to be discarded and if the line is instantaneously switched to zerodelay all of this information unless cancelled or predeleted, will bepresented in highly condensed form in the output signal. As a practicalmatter with conventional delay lines utilizing R and L or C componentsthere will be a time interval required for switching the line frommaximum to minimum delay and it has been found that even if the linedoes not contain signal information this switching of a line has asignificant minimum time constant associated with it which produces adisturbing transient audible in the output signal. The repetition rateof this transient is the reciprocal of the sample period. Because of thelimitations imposed by the parameters of the system as previously setforth herein, this switching frequency and spectral components of thetransient itself will always be within the audiorange and thus presentas a highly disagreeable intermoduation component in the audio output ofthe device. The present invention provides a number of implementationsfor transient suppression and message gap bridging arrangements for thepurpose of minimizing the disagreeable noise effects involved. In moreelaborate systems the substitution of pseudo or real message componentsfurther improves the transition from one sample to the next and can beadapted to fill in a portion of what is discarded in the compressionprocess.

Referring now to FIG. 4 a portion of the 333 Hz wave at the transitionpoint illustrated in FIG. 1(d) has been reproduced in which cycle 4 andcycle 9 of the original recorded 333 Hz wave are shown as a smoothuninterrupted sine wave. The junction between the end of cycle 4 and thebeginning of cycle 9 at point 41, although shown as a continuous portionof the sine wave, is in actuality, as previously stated, almost never sorelated in the non-selective periodic sampling of independent complexwaveforms and thus instead of a smooth transition point 41 a disjunctionbetween the end of one chunk and the beginning of the next chunk insuccessive samples is to be expected. This disjunction could undoubtedlybe accommodated with no loss of intelligibility if the transient fromswitching the line (either loaded or unloaded) did not have to be dealtwith at exactly this point in time. Since this transient is responsiblefor a highly annoying audible output from the system it must beeliminated and for this purpose a gating signal as indicated in FIG. 4(b) may be applied symmetrically with respect to the transition point 41to produce the output signal shown in FIG. 4(c). By making the gate longenought to encompass the transient resulting from switching the line,the audible noise so generated is eliminated. The improvement obtainedby this expedient, while significant, is not ideal since theintroduction of the gate signal within the audio range is itself audibleas a repetitive disjunctive gap which intermodulates with the audiosignal. This effect can be reduced by using an output filter designedfor the particular repetition rate and gate width to smooth the abrupttransition shown in FIG. 4(c) and this output response is indicated inFIG. 4(d).

A further improvement is possible by using the gating signal as a gaincontrol signal and tapering the off and perhaps the on transitions ofthe gate so that a gradual transition of the audio output from off to onis accomplished and a relatively smooth transition as indicated in FIG.4(f) results. The object is to mini mize the gap effect which in itselfhas an audio characteristic and can act like a cue. Tapering thetrailing edge of the gate helps this considerably whereas ananticipating start (or'relative delay of the speech signal) would bepreferable for gradual onset for the leading edge. With these relativelysimple expedients the smooth transition between adjacent chunks whichare disjunctively joined by the operation of the compression-discardprocess are achieved in a manner which is satisfactory for manyapplications.

Referring now to FIG. 5, the more elaborate arrangements for bridgingthe gap between adjacent samples will be described. As shown in FIG.5(a) a disjunctive transition which is the norm to be expectedrepresents a sharp discontinuity in the message signal and hassuperposed thereon the noise transient from switching the line aspreviously described. By introducing a gate signal FIG. 5(b) ofsufficient width to encompass the line switching transient andconditioning the gate to coincide with a zero level and same directionof change for the adjacent signals being processed a zero level gatingtransition as shown in FIG. 5(c) can be achieved. This transition whichis free of line switching noise and essentially continues an existingzero amplitude signal level during the interval of the gate has beenfound to provide little or no disturbance to the average listener. I

Because of the nature of the human hearing phenomenon, particularly theability of the'ear to synthesize the message it is concentrating uponeven in the presence of noise, it may be useful in certain circumstancesto introduce a pseudo or real message component in the zero levelinterval indicated in FIG. (c). For this purpose suitably selected noiseor signal components of approximately the same amplitude and frequencycan be inserted in what is otherwise a quiet gap interval in the messagestream and this arrangement of the invention is indicated in FIG. 5(d).Where the gap is to be filled with noise components, a suitable sourceand symmetrical switching to introduce noise from the source into thesignal channel can be readily applied during the gating interval.

FIG. 6 represents a preferred form of gap filling where two signalcontrolled delay lines are used. The speech signal is applied to bothdelay lines designated channel A and channel B in FIGS. 6(a) and 6(b)respectively and these two lines are signal controlled to havesymmetrical complementary gain characteristics and overlapping variabledelay characteristics as shown in FIGS. 6(c) and 6(d). Here the delaycontrol signals as shown in FIG. 6(d) are phased to overlap at least anamount corresponding to the transition portion of the gain controlcharacteristics of FIG. 6(c). The outputs of both delay channels A and Bare combined to produce the combined output shown in FIG. 6(c).

Generally the length of the delay lines used for channels A and B inFIG. 6 will employ one full length delay line and one relatively shorterlength delay line for stor ing the signal used for gap filling purposes.This arrangement will reduce the cost of the equipment represented bythe multiple section delay lines necessary to obtain the requiredmaximum delay length for system performance requirements. On the otherhand, for systems where cost is not a primary factor, two equal fulllength variable delay lines can be employed and their control signalscan be alternately applied so that the signal channel is through firstone and then the other delay line thereby giving a full signal periodfor switching the inactive delay line back to minimum delay conplied asindicated in FIG. 6(c).

Referring now to FIG. 7, a basic speech compressionexpansion system inaccordance with the invention will be described. This system comprises avariable speed playback device 51 which is indicated to be a tapetransport with a manual select speed control input 52. The signalderived from transporting the tape past a magnetic transducer is appliedto an AGC amplifier 53 whichalso passes the signal through a band passfilter having an adjustable low and high frequency cutoff. The selectionof the cutoff frequencies for the filter may be operated from manualcontrol 52 in conjunction with the selection of playback speed for theplayback device 51. The manual control-52 also supplies an amplitudecontrol signal to a fine voice pitch adjust control 54 which supplies online 55 a signal to control the end amplitude of the linearly increasingwaveform which controls the variable delay line as hereinafterdescribed.

The signal after passing through the amplifier and filter 53 enters avariable delay line 56 which can be signal controlled between minimumand maximum delay limits. This control signal applied on line 57 isderived from a ramp level and amplitude charger 58 which receives as itsinput either a compression triangular waveform on line 59 or anexpansion inverse of the waveform on line 59 which appears on line 61after passing through an inverter 62. One or the: other of the lines 59and 61 is energized with a ramp waveform depending upon the setting of aswitch 63 which supplies the basic ramp waveform from ramp pulse: traingenerator 64. The repetition period of the ramp waveform is selectableby a manual control 65. A pulse coincident with the reset of the linearportion of the ramp waveform appears on line 66 and is applied to ablanking pulse generator 67 to produce a blanking pulse output the widthof which can be controlled by manual adjustment 68 and which issynchronized with the input pulse on line 66.

The output of the variable delay line 56 is applied to a blankingcircuit and amplifier 71 which transmits or blocks the signal dependingupon the blanking pulse B applied on line 72 from generator 67 and whenthe blanking pulse is not present F the delay signal is applied to aspeech bandpass filter 73 the output of which is applied to an audioreproducer 74. y

In addition to the amplitude excursion established for the linearvoltage ramp signal from generator 64 which is controlled by manualcontrol 52 the absolute level of the voltage applied can be controlledby level adjust means 60. The variable delay line 56 will generally beof any known type and in particular may be 360 RC filter stages wherethe shunt resistor is provided by a PET or other semiconductor devicewhich varies resistance in response to a controlled voltage or current.Such delay lines generally perform best: with respect to distortion ofthe signal passing therethrough if the phase delay per stage is keptwell below the maximum possi ble value of 90. Accordingly, the line canbe designed to operate with 45 to 60 phase delay per stage maximum andthe number of stages is then determined as greater than the quantity: N(6 or 8) c(f,,, AT In the above inequality the digits o and 8 representthe number of stages per electrical cycle of the highest frequency to bepassed corresponding to a phase delay of 60 or 45, respectively, asthemaximum phase shift per stage which is to be utilized; the quantity cis the compression ratio; the quantity f is the highest frequency beingpassed by the line; and AT is the maximum signal delay desired asdictated by the maximum permissible discard interval previouslyspecified. Many other forms of delay line construct-ions which arecapable of being signal controlled are known in the art and the presentinvention is not to be considered as limited to any particular form ofdelay line.

Referring now to FIGS. 8(a) and 8(b) the operation of the system of FIG.7 will be described. The sample period waveform 81 has an adjustableperiod set by control for producing an asymmetrical sawtooth waveform 82which produces a relatively long negative. going linear voltage followedby a shorter positive going linear voltage. This waveform is useddirectly on line 59 for speech compression while, after inversion ininverter 62, its inverse is used on line 61 for expansion. The expansionwaveform is indicated in dotted lines at 83 inFIG. 8(a). For a variabledelay line 56 which increases delay as the control voltage becomes morenegative, the waveforms 82 and 83 have the proper sense for controllingthe delay interval and the magnitude of the delay is determined byamplitude control 52 relative to a voltage level set by the level adjust60. Thus 'the operating point in the excursion of the waveform 82 isselected for a given compression ratio in conjunction with the sampleperiod which will be a predetermined combination for any givencompression ratio assuming the maximum delay AT in the line 56 is afixed value as obtained by selecting line length according to the valued-T as given in FIG. 3(a) and Table I for the desired compression ratio.If the maximum delay to the signal is not maintained constant thediscard period will change correspondingly as is evident from thedescription of FIG. 1 and corresponding adjustments in the amplitude ofthe wave will be required to give the slope (I required for acompression ratio 0. Similar considerations apply for the slope of curve83 which must be set at its corresponding value d for an expansion ratioe.

The operation of blanking pulse generator 67 is shown to produce a pulse84 in FIG. 8(b) of predetermined width in response to the start pulse ofthe sample period signal 81 received on line 66. This pulse may beapplied in gain control fashion to the circuit 71 with modified trailingedge as previously described to reduce the transient signal and providea gradual onset of voice sound signals which are passed to thetransducer 74. The width B of the blanking pulse is selected withcontrol 68 and is normally made of sufficient duration to permit theshort steep linear-portion of the ramp waveform to return the delay line56 to its zero or minimum delay condition and dissipate the signalenergy therein (or the transient caused by switching the line itself)prior to enabling the signal. channel which energizes the transducer 74with the subsequent speech signal segments.

The blanking period B and enabled period B for expansion mode are shownin FIG. 8(c). The expanded chunks with an initial output gap are shownin FIG. 8(d).

The system of FIG. 7 can also be used to substitute noise or pseudosignal gap filling signals corresponding to the system described in FIG.5. For this purpose a source 75 of such signals is arranged to supplythe input signal to filter 73 during the blanking interval. By means ofa switch 76 this gap filling during the blanking interval can be madeoptional. The gap filling signal 75 can also be derived from the messagesignal output of amplifier 53.

Referring now to FIG. 9, a modified-form of the invention particularlysuitable for accomplishing the various gap filling procedures for thespeech compression systems previously described will be disclosed.Portions of FIG. 9 which are essentially the same as those described inFIG. 7 have corresponding reference numerals and accordingly only theadditions and changes will be further described. In addition to thevariable delay line 56 a second variable delay line 91 receives thesignal wave from amplifier 53. The output of the delay lines 56 and 91are applied respectively to complementary blanking circuits 92 and 93.Signals passed by these blanking circuits 92 and 93 are amplified andfiltered in element 73 and passed to the acoustic reproducer 74 asheretofore described.

A pulse train generator 94 produces a pulse wave train as shown in FIG.10(a) having a selectable pulse repetition rate determined by thesetting of manual control thereby establishing the basic sample period.The output pulse, from generator 94 is delayed in delay unit 95 andapplied to a first ramp generator 96 and in undelayed form is applied toa second ramp generator 97. The ramp generators 96 and 97 are subject towaveform level control from manual adjust element 60 and ramp linearwave amplitude control from the manual adjust element 52. As previouslystated, the fine pitch adjustment 54 may be provided for slightlymodifying the ramp slope as a voice pitch adjustment by effectivelyaltering the frequency conversion over a small range. In addition theblanking width interval of each generator is adjustable with controls 68and 70 respectively. The outputs of the ramp generators 96 and 97 areapplied respectively to delay lines 56 and 91 to control the time delayof signals passing through the respective lines in accordance with thecontrol signals applied. By means of c or e select controls the sense ofthe slope of the ramp waveforms can be selected for compression orexpansion.

The level and amplitude controls for setting the respective rampgenerators 96 and 97 are preferably relatively adjustable to permitselection of the relationship between the two ramp waveforms. By makingthe delay and phasing of the unit 95 adjustable any desired delay lineoverlap can also be achieved. It is also possible to rearrange thecomponents to have the complementary gating at the inputs of the twodelay lines 56 and 91 with the outputs switched to be combined in acommon channel to amplifier 73. This alternative discards the portion ofthe speech signal that is not utilized by each line before it enters theline and thus eliminates the necessity for dissipating these portionswhen the lines are switched between active periods.

Referring now to FIG. 10, the operation of the speech compression systemof FIG. 9 will be described. The pulse train generator 94 produces thetiming waveform of FIG. 10(a). This pulse triggers the transition ofwaveform C2 in pulse ramp generator 97 which produces the blanking pulseindicated in FIG. 10(0) with the predetermined width of B and B beingdetermined by the blankingpulse width control 68. After the delayindicated in FIG. 10(b) the pulse from generator 94 triggers the rampgenerator 96 to produce the waveform Cl shown in FIG. 10(1)). With thisarrangement the control wave CI for the delay line 56 is overlapped intime by waveform C2 having slope in the same sense and bridging thesteep return slope waveform of ramp wave Cl. With the asymmetrical timeintervals shown in FIG. 10, the arrangement for gap filling modes ofoperations shown in FIGS. 5 and 6 can be practiced. By making thewaveforms C1 and C2 have symmetrical rising and falling portions thearrangement is suitable for alternate switching of the lines 56 and 91to provide alternate compressed (or expanded) chunks of the speechsample. The choice of the relative lengths of sample through line 56 and91 will generally be dictated by manufacturing costs for the delay line.Thus for a main delay line 56 of adequate length for the compressionratio desired, a relatively shorter line 91 used only for gap fillingpurposes will generally be more economical. On the other hand, two fulllength lines which are alternately active to pass speech sample chunksthereby providing adequate time for the non-active line to be returnedto its minimum delay condition will provide for smooth transitions, anydesired overlap and the maximum time interval for discharging the lineto minimum delay condition prior to its processing the next speechsample. The action of the system of FIG. 9 in the gap filling mode isindicated in FIG.- 10(d) and generally corresponds to that previouslydescribed with respect to FIG. 5.

The operation of the system of FIG. 9 for speech expansion, i. e.,increasing the time duration for a given speech utterance and increasingthe frequency components thereof from a reproducer running at a slowerthan recorded rate is shown in FIG. 1 1. Here the ramp generators 96 and97 have inverted outputs to produce the expansion waveforms E1 and E2shown in FIGS. 11(a) and 11(c) respectively and the blanking waveformhas been made symmetrical such that the delay lines 56 and 91 are usedalternately for approximately equal periods. By the nature of speechexpansion, a gap in the signal output will always occur since the linesare controlled to change from maximum delay at the start of the sampleto minimum to zero delay at the end of the sample. Thus when the line isswitched to maximum delay there will inevitably be a time gap beforedelayed signal emerges from the output end of the line. Applying thecontrol sequence indicated at FIG. 11 the speech samples processed bylines 56 and 91 are over lapped so as to fill the gap as indicated inFIG. 11(d) by the solid and dotted outlined signal chunks E and E Thepresence of a slight overlap in the reproduced signal does notsignificantly interfere with intelligibility since it generally is notnoticeable and at worst may result in a slight echo effect of the typecommonly encountered in a telephone conversation. The time expandedspeech waveform obtained using the mode of operation indicated in FIG.11 is useful for the recognition and comprehension of difficult passagesand for analysis and study of foreign languages and the like.

The system shown in FIG. 12 represents a simplification of the system ofFIG. 9 where a fixed delay line 101 is used in place of the secondvariable delay line 91 of FIG. 9. The control of blanking ciruits 92',93' is simplified in that the variable width blanking gate B as derivedfrom pulse train generator 94 correspondingly produces gaps in theoutput signals which have been delayed by passage through variable delayline 56. The fixed delay of line 101 is selected to further delay someportion of the signal emerging from the delay line 56 by an amountsufficient to fill the gap caused by blanking pulse B therebyessentially repeating some portion of each message chunk while thevariable delay line 56 is switched back to its minimum delay condition.Again, this repetition is not objectionable and may merely introduce aslight echo effect which is much less objectionable than the presence ofthe gap in the message signal. This sequence of operation is shown inFIG. 13 where the variable chunk C and the fixed chunk C alternate insupplying the output.

' The expansion mode for operation of the circuit of FIG. 12 is shown inFIG. 14 where the ramp signals are inverted for the expansion waveformwhich controls the delay line 56 to vary from maximum delay to minimumdelay over the linear ramp portion E shown in FIG. 14(a). The blankingwaveform B is selected to pass some portion of the signal chunk throughthe appropriate amount of delay to fill the gap between chunks in theoutput as indicated in'FIG. 14(c). Thus the output is composed of chunksE and E in alternation for continuous signal.

The system of FIG. 12 could be further simplified by eliminating delayline 101 and conditioning gate 93' to introduce in the gap interval'anypseudo or noise signal from a suitable source which would simulate thefrequency content of the actual speech signal. While this version wouldbe less desirable than using the actual speech signal for gap filling itwould, nevertheless, be better than reproducing the speech signal withthe message gaps present since the audible effect of gaps becomesdetrimental to recognition of the message content, especially at highcompression ratios. This modification gives a mode of operation similarto the optional noise gap filling described for FIG. 7.

FIG. 15 shows a modification of the invention for binaural processing.The speech signal from the band pass filter 53 is applied to symmetricalvariable delay lines VDLl and VDLZ controlled by waveform generator 102.The output of VDLl is applied as an input to gates 103 and 105. Theoutput of VDL is applied as an input to gates 104 and 106. The delayline VDL, is con trolled for linear variation of delay according to thewaveform of FIG. 16(c). The delay line VDL is controlled for linearvariation of delay according to the waveform of FIG. 16(d). Each ofthese waveforms has its rapid return transition at the mid-point of thelinear delay portion of the other waveform.

The gates 1t 3 and 106 are controlled by gating waveforms B and 13,shown in FIG. 16(e). Gate 103 passes signal during B and is blockedduring B Gate 106 is blocked during B and passes signal during B,Amplifier 107 combines the outputs of gates 103 and 106 and applies thecombined signal to an audio reproducer 108.

The gates 104 and are controlled by the gating wavefonns B and B, shownin FIG. 16(f). Gate 104 passes signal during fi gnd is blocked during BGate 105 is blocked during B and passes signal during B Amplifier 109combines the outputs of gates 104 and 105 and applies the combinedsignal to an audio reproducer l 10.

The system of FIG. 15 operates to reproduce the entire original signal(for compression ratio equal to two) since each delay line processes theportion which is the discard for the other line as is evident from FIGS.16(a) and 16(b). For compression ratios greater than two some messagediscard occurs and for ratios less than two the overlap or messageduplication increases in the output. By listening binaurally, however,the intelligibility is enhanced since the overall discard is eliminated(or greatly reduced for the higher compression ratios) and the overlapor repeat of message portions is not detrimental to word detection bythe listener.

A binaural system without supplemental gap filling (as just described)would be achieved by removing gates 105 and 106 in FIG. 15. The linesVDL, and VDL, would supply the processed signal in alternation to therespective output transducers 108 and 110 for binaural output.

FIG. 17 shows the invention using a form of delay line capable ofprocessing speech signals in a manner which greatly reduces the problemsassociated with dis carding stored information in the line. The; systemshown in FIG. 17 comprises an analog shift register having a pluralityof stages ASR ASR ASR,,, which has a speech signal input applied on line111 and a compressed or expanded speech signal output on line 112.Alternate stages of the delay line are clocked by

1. A processor for electric signals representing coded audible signalssuch as speech or the like comprising: means for deriving said electricsignals as analog representations of said audible signals with thefrequency components of said electric signals related by a given factorto the frequency components of said audible signals; variable delay linemeans having an input and an output, said input coupled to said meansfor deriving said electric signals for propagating representations ofsaid electric signals to said output with controllable time delay; meansfor controlling said delay line means for periodic linear variation ofsaid time delay between predetermined delay values; and means coupled tosaid output of said delay line means and responsive substantially onlyto signals propagating through said delay line means which have beensubject to unidirectional variation of said time delay for producingoutput signals reproducible as an audible representation of saidelectric signals having frequency components altered by substantiallysaid factor to approximate the frequency coMponents of said audiblesignals.
 2. A processor for electric signals representing coded audiblesignals such as speech or the like comprising: means for deriving saidelectric signals as analog representations of said audible signals withthe frequency components of said electric signals related by a givenfactor to the frequency components of said audible signals; variabledelay line means having an input and an output, said input coupled tosaid means for deriving said electric signals for propagatingrepresentations of said electric signals to said output withcontrollable time delay; means for controlling said delay line means forperiodic linear variation of said time delay between predetermined delayvalues; and means coupled to said output of said delay line means andresponsive substantially only to signals propagating through said delayline means which have been subject to unidirectional variation of saidtime delay for producing output signals reproducible as an audiblerepresentation of said electric signals having frequency componentsaltered by substantially said factor to approximate the frequencycomponents of said audible signals; and in which said means forcontrolling said delay line means provides periods of linearly changingunidirectional delay variation alternating with reset intervals whereinsaid delay line is returned to an initial predetermined delay value andincluding means for blanking output signals from said output of saiddelay line means to substantially zero amplitude during said resetintervals.
 3. Apparatus according to claim 2 in which said means forblanking includes means for gradually restoring the amplitude of saidoutput signals to full amplitude after said reset intervals. 4.Apparatus according to claim 1 and including input high pass filtermeans, said filter means substantially eliminating components of saidelectrical signals from propagating through said delay line means whichare below a frequency approximately equal to said factor times thereciprocal of the period of said linear variation.
 5. Apparatusaccording to claim 4 and including band pass output filter means forlimiting said output signals to a predetermined audio bandwidth. 6.Apparatus according to claim 1 and including: input high pass filtermeans having adjustable low frequency cutoff interposed between saidmeans for deriving said electric signals and said input of said variabledelay line means; means for adjusting the effective rate of saidperiodic linear variation to adjust said factor by which frequenciespropagated through said delay line are altered; and means coupling saidcutoff adjusting means and said effective rate adjusting means forrelated variation to eliminate low frequency components which wouldotherwise produce frequency components in said delay line means below anaudio low frequency limit related to said factor and the period of saidvariation.
 7. Apparatus according to claim 6 in which said means forcontrolling said delay line means provides periods of linearly changingunidirectional delay variation alternating with reset intervals whereinsaid delay line is returned to an initial predetermined delay value andincluding means for blanking output signals from said output of saiddelay line means to substantially zero amplitude during said resetintervals.
 8. Apparatus according to claim 7 in which said means forblanking includes means for gradually restoring the amplitude of saidoutput signals to full amplitude after said reset intervals. 9.Apparatus according to claim 2 in which said means for blanking includesmeans responsive to the level of said output signals being zero at thebeginning of said reset interval to initiate said blanking and meansresponsive to the level of said output signals being zero at the end ofsaid reset interval to terminate said blanking.
 10. Apparatus accordingto claim 2 and including means for introducing an audio signal segmenTinto for combining with said audible representation of said electricsignals during said reset interval.
 11. A processor for electric signalsrepresenting coded audible signals such as speech or the likecomprising: means for deriving said electric signals as analogrepresentations of said audible signals with the frequency components ofsaid electric signals related by a given factor to the frequencycomponents of said audible signals; variable delay line means having aninput and an output, said input coupled to said means for deriving saidelectric signals for propagating representations of said electricsignals to said output with controllable time delay; means forcontrolling said delay line means for periodic linear variation of saidtime delay between predetermined delay values; and means coupled to saidoutput of said delay line means and responsive substantially only tosignals propagating through said delay line means which have beensubject to unidirectional variation of said time delay for producingoutput signals reproducible as an audible representation of saidelectric signals having frequency components altered by substantiallysaid factor to approximate the frequency components of said audiblesignals; and in which said means for controlling said delay line meansprovides periods of linearly changing unidirectional delay variationalternating with reset intervals wherein said delay line is returned toan initial predetermined delay value and further including: second delayline means coupled to said means for deriving said electric signals; andmeans for interrupting signals from the output of said variable delayline means and substituting signals from the output of said second delayline means during said reset intervals to produce said output signalsreproducible as said audible representation.
 12. Apparatus according toclaim 11 in which said second delay line means include a variable delayline and including means for controlling said second delay line meansfor periodic linear variation of said time delay between predetermineddelay values, said variation of said second delay line means occurringduring said reset intervals for the first recited said delay line means.13. A processor for electric signals representing coded audible signalssuch as speech or the like, said electric signals being analogrepresentations of said audible signals with the frequency components ofsaid electric signals related by a given factor to the frequencycomponents of said audible signals comprising: first and second variabledelay line means coupled to a source of said electric signals forpropagating said electric signals therein with controllable time delay;means for controlling said first and second delay line means forperiodic linear variation of said time delay between predetermined delayvalues, said means for controlling said first and second delay linemeans providing periods of linearly changing unidirectional delayvariation alternating with reset intervals wherein said delay lines arereturned to an initial predetermined delay value, said reset intervalsfor each said delay line means occurring within the period of theunidirectional delay variation of the other said delay line means; andmeans coupled to the outputs of said first and second delay line meansand responsive substantially only to signals propagating through saiddelay line means which have been subject to unidirectional variation ofsaid delay for producing an audible representation of said electricsignals having frequency components altered by substantially said factorto approximate the frequency components of said audible signals. 14.Apparatus according to claim 13 in which said first and second delayline means vary between the same said predetermined delay valuesalternately with substantially equal periods of said linear variation.15. Apparatus according to claim 14 in which said reset intervals aresubstantialLy shorter than the periods of said linear variation andoccur for each said delay line means approximately at the mid-point ofsaid linear variation for the other said delay line means and including:separate audio reproducer means coupled to the respective outputs ofsaid first and second delay line means for producing simultaneouslyseparate versions of said audible representation from each saidreproducer means.
 16. Apparatus according to claim 11 and including:input high pass filter means having adjustable low frequency cutoffinterposed between said means for deriving said electric signals andsaid input of said variable delay line means; means for adjusting theeffective rate of said periodic linear variation to adjust said factorby which frequencies propagated through said delay line are altered; andmeans coupling said cutoff adjusting means and said effective rateadjusting means for related variation to eliminate low frequencycomponents which would otherwise produce frequency components in saiddelay line means below an audio low frequency limit related to saidfactor and the period of said variation.
 17. Apparatus according toclaim 16 in which said second delay line means is also a variable delayline and including means for controlling said second delay line meansfor periodic linear variation of said time delay between predetermineddelay values, said variation of said second delay line means occurringduring said reset intervals for the first recited said delay line means.18. A processor for electric signals representing coded audible signalssuch as speech or the like, said electric signals being analogrepresentations of said audible signals with the frequency components ofsaid electric signals related by a given factor to the frequencycomponents of said audible signals comprising: first and second variabledelay line means coupled to a source of said electric signals forpropagating said electric signals therein with controllable time delay;input high pass filter means having adjustable low frequency cutoffinterposed between said source and the input of said variable delay linemeans; means for controlling said first and second delay line means forperiodic linear variation of said time delay between predetermined delayvalues, said means for controlling said first and second delay linemeans providing periods of linearly changing unidirectional delayvariation alternating with reset intervals wherein said delay lines arereturned to an initial predetermined delay value, said reset intervalsfor each said delay line means occurring within the period of theunidirectional delay variation of the other said delay line means; meansfor adjusting the effective rate of said periodic linear variation toadjust the factor by which frequencies propagated through said delayline are altered; means coupling said cutoff adjusting means and saideffective rate adjusting means for related variation to eliminate lowfrequency components which would otherwise produce frequency componentsin said delay line means below an audio low frequency limit related tosaid factor and the period of said variation; and, means coupled to theoutputs of said first and second delay line means and responsivesubstantially only to signals propagating through said delay line meanswhich have been subject to unidirectional variation of said delay forproducing an audible representation of said electric signals havingfrequency components altered by substantially said given factor toapproximate the frequency components of said audible signals. 19.Apparatus according to claim 18 in which said first and second delayline means vary between the same said predetermined delay valuesalternately with substantially equal periods of said linear variation.20. Apparatus according to claim 19 in which said reset intervals aresubstantially shorter than the periods of said linear variation andoccur for each said delay linE means approximately at the mid-point ofsaid linear variation for the other said delay line means and including:separate audio reproducer means coupled to the respective outputs ofsaid first and second delay line means for producing simultaneouslyseparate versions of said audible representation from each saidreproducer means.
 21. Apparatus according to claim 11 in which saidsecond delay line means is a fixed delay line the input of which iscoupled to an output of said variable delay line means.
 22. Apparatusaccording to claim 21 and including: input high pass filter means havingadjustable low frequency cutoff interposed between said source and theinput of said variable delay line means; means for adjusting theeffective rate of said periodic linear variation to adjust said factorby which frequencies propagated through said delay line are altered; andmeans coupling said cutoff adjusting means and said effective rateadjusting means for related variation to eliminate low frequencycomponents which would otherwise produce frequency components in saiddelay line means below an audio low frequency limit related to saidfactor and the period of said variation.
 23. A processor for recordingsrepresenting coded audible signals such as speech or the likecomprising: playback means operable to reproduce said recording at arate faster by a given factor than the recording rate thereby producingelectric signals which are the time compressed analog of said audiblesignals with frequency components increased by said factor; variabledelay line means coupled to said playback means for propagating saidelectric signals therein with controllable time delay; means forcontrolling said delay line means for periodic linear increase of saidtime delay between predetermined delay values; and, means coupled to theoutput of said delay line means and responsive subtantially only tosignals propagating through said delay line means which have beensubject to said increase of said time delay for producing an audiblerepresentation of said electrical signals having frequency componentsreduced substantially by said factor to approximate the frequencycomponents of said audible signals.
 24. Apparatus according to claim 23and including input high pass filter means, said filter meanssubstantially eliminating components for said electrical signals frompropagating through said delay line means which are below a frequencyapproximately equal to said factor times the reciprocal of the period ofsaid linear variation.
 25. Apparatus according to claim 24 and includingband pass output filter means limiting said output representation to apredetermined audio bandwidth.
 26. Apparatus according to claim 23 andincluding: input high pass filter means having adjustable low frequencycutoff interposed between said playback means and the input of saidvariable delay line means; means for adjusting the effective rate ofsaid periodic linear variation to adjust said factor by whichfrequencies propagated through said delay line are altered; and meanscoupling said cutoff adjusting means and said effective rate adjustingmeans for related variation to eliminate low frequency components whichwould otherwise produce frequency components in said delay line meansbelow an audio low frequency limit related to said factor and the periodof said variation.
 27. Apparatus according to claim 26 in which saidmeans for controlling said delay line means provides periods of linearlychanging unidirectional delay variation alternating with reset intervalswherein said delay line is returned to an initial predetermined delayvalue and including means for blanking output signals from said outputof said delay line means to substantially zero amplitude during saidreset intervals.
 28. Apparatus according to claim 27 in which said meansfor blanking includes means for gradually restoring the amplitude ofsaid output to full amplitude aFter said reset intervals.
 29. Apparatusaccording to claim 2 in which said means for blanking includes meansresponsive to the level of said output signals being zero at thebeginning of said reset interval to initiate said blanking and meansresponsive to the level of said output signals being zero at the end ofsaid reset interval to terminate said blanking.
 30. Apparatus accordingto claim 29 and including means for introducing an audio signal segmentinto said output signals and said audible representation of saidelectric signals during said reset interval.
 31. Apparatus according toclaim 23 in which said means for controlling said delay line meansprovides periods of linearly changing unidirectional delay variationalternating with reset intervals wherein said delay line is returned toan initial predetermined delay value and including: second delay linemeans coupled to a source of said electric signals; and means forinterrupting signals from the output of said variable delay line meansand substituting signals from the output of said second delay line meansduring said reset intervals to produce said audible representation. 32.Apparatus according to claim 31 in which said second delay line meansincludes a variable delay line and including means for controlling saidsecond delay line means for periodic linear variation of said time delaybetween predetermined delay values, said variation of said second delayline means occurring during said reset intervals for the first recitedsaid delay line means.
 33. A processor for recordings representing codedaudible signals such as speech or the like comprising: playback meansoperable to reproduce said recording at a rate faster by a given factorthan the recording rate thereby producing electric signals which are thetime compressed analog of said audible signals with frequency componentsincreased by said factor; first and second variable delay line meanscoupled to said playback means for propagating said electric signalstherein with controllable time delay; means for controlling said firstand second delay line means for periodic linear variation of said timedelay between predetermined delay values, said means for controllingsaid first and second delay line means providing periods of linearlychanging unidirectional delay variation alternating with reset intervalswherein said delay lines are returned to an initial predetermined delayvalue, said reset intervals for each said delay line means occurringwithin the period of the unidirectional delay variation of the othersaid delay line means; and means coupled to the outputs of said firstand second delay line means and responsive substantially only to signalspropagating through said delay line means which have been subject tounidirectional variation of said delay for producing an audiblerepresentation of said electric signals having frequency componentsreduced by substantially said factor to approximate the frequencycomponents of said audible signals.
 34. Apparatus according to claim 33in which said first and second delay line means vary between the samesaid predetermined delay values alternately with substantially equalperiods of said linear variation.
 35. Apparatus according to claim 34 inwhich said reset intervals are substantially shorter than the periods ofsaid linear variation and occur for each said delay line meansapproximately at the mid-point of said linear variation for the othersaid delay line means and including: separate audio reproducer meanscoupled to the respective outputs of said first and second delay linemeans for producing simultaneously separate versions of said audiblerepresentation from each said reproducer means.
 36. Apparatus accordingto claim 31 and including: input high pass filter means havingadjustable low frequency cutoff interposed between said source and theinput of said variable delay line means; means for adjusting theeffective rate of said pEriodic linear variation to adjust said factorby which frequencies propagated through said delay line are altered; andmeans coupling said cutoff adjusting means and said effective rateadjusting means for related variation to eliminate low frequencycomponents which would otherwise produce frequency components in saiddelay line means below an audio low frequency limit related to saidfactor and the period of said variation.
 37. Apparatus according toclaim 36 in which said second delay line means is also a variable delayline and including means for controlling said second delay line meansfor periodic linear variation of said time delay between predetermineddelay values, said variation of said second delay line means occurringduring said reset intervals for the first recited said delay line. 38.Apparatus according to claim 18 in which said source of said electricsignals comprises playback means operable to reproduce a recording at arate faster by a given factor than the recording rate, thereby producingelectric signals which are the time compressed analog of said audiblesignals with frequency components increased by said given factor. 39.Apparatus according to claim 18 in which said first and second delayline means vary between the same said predetermined delay valuesalternately with substantially equal periods of said linear variation.40. Apparatus according to claim 39 in which said reset intervals aresubstantially shorter than the periods of said linear variation andoccur for each said delay line means approximately at the mid-point ofsaid linear variation for the other said delay line means and including:separate audio reproducer means coupled to the respective outputs ofsaid first and second delay line means for producing simultaneouslyseparate versions of said audible representation from each saidreproducer means.
 41. Apparatus according to claim 40 in which saidsecond delay line means is a fixed delay line the input of which iscoupled to an output of said variable delay line means.
 42. Apparatusaccording to claim 41 and including: input high pass filter means havingadjustable low frequency cutoff interposed between said source and theinput of said variable delay line means; means for adjusting theeffective rate of said periodic linear variations to adjust said factorby which frequencies propagated through said delay line are altered; andmeans coupling said cutoff adjusting means and said effective rateadjusting means for related variation to eliminate low frequencycomponents which would otherwise produce frequency components in saiddelay line means below an audio low frequency limit related to saidfactor and the period of said variation.
 43. A processor for recordingsrepresenting coded audible signals such as speech or the likecomprising: playback means operable to reproduce said recording at arate slower by a given factor than the recording rate thereby producingelectric signals which are the time expanded analog of said audiblesignals with frequency components decreased by said factor; variabledelay line means coupled to said playback means for propagating saidelectric signals therein with controllable time delay; means forcontrolling said delay line means for periodic linear decrease of saidtime delay between predetermined delay values; and means coupled to theoutput of said delay line means and responsive substantially only tosignals propagating through said delay line means which have beensubject to said decrease of said time delay for producing an audiblerepresentation of said electrical signals having frequency componentsincreased substantially by said factor to approximate the frequencycomponents of said audible signals.
 44. Apparatus according to claim 43in which said means for controlling said delay line means providesperiods of linearly changing unidirectional delay variation alternatingwith reset intervals wherein said delay line Is returned to an initialpredetermined delay value and including: second delay line means coupledto a source of said electric signals; and means for interrupting signalsfrom the output of said variable delay line means and substitutingsignals from the output of said second delay line means during saidreset intervals to produce said audible representation.
 45. Apparatusaccording to claim 44 in which said second delay line means include avariable delay line and including means for controlling said seconddelay line means for periodic linear variation of said time delaybetween predetermined delay values, said variation of said second delayline means occurring during said reset intervals for the first recitedsaid delay line means.
 46. Apparatus according to claim 44 in which saidsecond delay line means is a fixed delay line the input of which iscoupled to an output of said variable delay line means.
 47. Apparatusaccording to claim 1 in which said delay line means is controlled suchthat: the time delay difference between said delay values is d X T whered 2 c-1/c+1, c is said factor and T is said period; and the period ofsaid variation is greater than the period of the lowest input frequencycomponent to be processed through said line divided by said factor. 48.Apparatus according to claim 1 in which: said factor is a number greaterthan one for time compression and less than one for time expansion; theperiod of said variation is greater than the period of the lowestfrequency component of said electric signals appearing at said output ofsaid delay line; and said unidirectional variation of said delay isgiven by 2 (c-1/c+1)t where t is the time and c is said factor.
 49. Aprocessor according to claim 1 in which said variable delay line meansis an analog shift register and said means for controlling said delayline means is a variable frequency clock pulse generator having afrequency that periodically varies inversely with time.
 50. A processoraccording to claim 11 in which both of said delay line means are analogshift registers each having a variable frequency clock pulse generator.51. Apparatus according to claim 1 in which said variable delay linemeans is a parallel r-word-n-stage shift register and including inputA/D means for converting said electric signals into a succession ofr-bit-digital words; output D/A means for converting the digital wordfrom the n-th stage into an analog signal and variable frequency clocksignal means for transferring the r-bit words from stage to stagethrough said n stages to effect linear variation of time delay for saidsignal.
 52. Apparatus according to claim 1 in which said variable delayline means comprises input A/D converter means, serializer means toproduce a digital word sequence, a serial digital shift register,parallelizer means to produce a parallel digital word and a D/Aconverter means; said control means comprising a variable frequencyclock generator for said shift register.
 53. Apparatus according toclaim 1 in which said delay line means comprises an analog storagematrix and including write control means for entering said electricsignals into said matrix at a first predetermined clock rate; and readcontrol means for reading said electric signals from said matrix at adifferent second predetermined rate.
 54. Apparatus according to claim 1in which said variable delay line means is a digital storage matrixhaving A/D input means and D/A output means; and means for writingdigital information into storage in said matrix at one rate and readingstored information out of said matrix at another rate.
 55. The method ofprocessing random speech signals to convey information intelligibly to ahuman listener at a rate different than the normal speaking equivalentfor said information and without obJectionable alteration of thefrequency components of said equivalent as reproduced for the listenercomprising the steps of: developing a full speech signal train of saidinformation with the elapsed time for said signal train altered into atime interval differing by a predetermined factor from that of thenormal speaking equivalent for said signal train thereby changing bysaid factor the frequency of the spectral components in said signaltrain relative to said equivalent; processing said signal train by alinear periodic time delay function to alter a regular succession ofpredetermined length segments of said signal train into segmentsapproximating a continuous signal with the frequency components in saidcontinuous signal altered by said factor relative to said components insaid signal train to approximate the components of said normal speakingequivalent; and reproducing said continuous signal as an intelligiblerepresentation of said information content with the elapsed time alteredby said factor but with the frequency components unchanged relative tosaid equivalent.
 56. Apparatus according to claim 2 in which said meansfor blanking includes means responsive to said output signals forinitiating and terminating said blanking for zero output signal levelsand for signal excursion adjacent said zero levels in the samedirection.
 57. Apparatus according to claim 43, in which said means forcontrolling said delay line means provides periods of linearly changingunidirectional delay variation alternating with reset intervals whereinsaid delay line is returned to an initial predetermined delay value andincluding means for blanking output signals from said output of saiddelay line means to substantially zero amplitude during said resetintervals.
 58. A processor for electric signals representing codedaudible signals such as speech or the like, said electric signals beinganalog representations of said audible signals with the frequencycomponents of said electric signals related by a given factor to thefrequency components of said audible signals comprising: two similardelay lines coupled to a source of said electric signals for propagatingsaid electric signals between the input and output terminals thereofwith controllable frequency transformation; means for controlling saiddelay lines alternately for periodic propagation of said electricsignals to produce predetermined frequency transformation of signalsemerging at the outputs of said delay lines; and means coupled to saidoutputs of both said delay line means and responsive substantially onlyto signals propagating through said delay lines which have been subjectto said predetermined frequency transformation for producing an audiblerepresentation of said electric signals having frequency componentsaltered by substantially said factor to approximate the frequencycomponents of said audible signals.
 59. Apparatus according to claim 58in which said two delay lines are analog shift registers.
 60. Apparatusaccording to claim 58 in which said analog shift registers arecontrolled alternately by a clock frequency periodically varied betweenpredetermined limits.
 61. The method of processing random speech signalsto convey information intelligibly to a human listener at a ratedifferent than the normal speaking equivalent for said information andwithout objectionable alteration of the frequency components of saidequivalent as reproduced for the listener comprising the steps of:developing a full speech signal train of said information with theelapsed time for said signal train altered into a time intervaldiffering by a predetermined factor from that of the normal speakingequivalent for said signal train thereby changing by said factor thefrequency of the spectral components in said signal train relative tosaid equivalent; propagating said signal train through a storage channelwithin which it is subject to a periodic frequency transformAtionfunction to alter a regular succession of predetermined length inputsegments of said signal train into output segments approximating acontinuous signal with the frequency components in said continuoussignal altered by the inverse of said factor relative to said componentsin said signal train to approximate the components of said normalspeaking equivalent; and reproducing said continuous signal as anintelligible audible representation of the information content of saidsignal train but with the frequency components unchanged relative tosaid equivalent.
 62. The method according to claim 61 in which saidstorage channel comprises analog shift register means controlled by aclock frequency periodically varied between predetermined frequencyvalues.
 63. The method according to claim 62 in which said inputsegments are propagated alternately through separate analog shiftregisters for frequency transformation.
 64. Apparatus according to claim48 in which said delay line means comprises analog shift register meansand said periodic variation of said time delay is obtained bycontrolling the clock frequency for said shift register means to providesaid unidirectional variation of said delay to be: N (p-1/p) ( 1/ft-(1/fo ) 2 (c-1/c+1) t where ft is the clock frequency at time t, N isthe number of stages of phase p and fo is the initial clock frequency.65. In a processor for electric signals such as those representing thecoded audible sounds of speech or the like, said electric signals beinganalog representations of said audible sounds with the frequencycomponents of said electric signals related by a given factor to thefrequency components of said audible sounds, the improvement comprising:controllable delay means having an input and an output, said inputcoupled to a source of said electric signals for passing signals fromsaid source into said delay means, and said delay means passing signalstherein to said output with controllable time delay; means forcontrolling said delay means with repetitive variation of saidcontrollable time delay between predetermined delay values the period ofsaid variation being greater than the period of the lowest frequencycomponent of said electric signals at said output of said delay meansthereby progressively delaying signals during each period of saidrepetitive variation as they pass through said delay means to obtain apredetermined frequency transformation of said electric signals asdetermined by the relation: